Rate adaptation for use in adaptive multi-rate vocoder

ABSTRACT

The present invention includes a time-division-multiple-access (TDMA) communication system having a base station and at least one mobile station, each transmitting and receiving an analog radio-frequency signal carrying digitally coded speech. The speech is encoded using a vocoder which samples a voice signal at variable encoding rates. During periods when the radio-frequency channel is experiencing high levels of channel interference, the encoded voice channel having a lower encoding rate is chosen. This low-rate encoded voice is combined with the high degree of channel coding necessary to ensure reliable transmission. When the radio-frequency channel is experiencing low levels of channel interference, less channel coding is necessary and the vocoder having a higher encoding rate is used. The high-rate encoded voice is combined with the lower degree of channel coding necessary to ensure reliable transmission. The appropriate levels of channel coding necessary for reliable transmission are determined by various channel metrics, such as frame erase rate and bit error rate. The determination of the appropriate vocoder rate and level of channel coding for both the uplink and downlink may be determined centrally at the base station, with the vocoder rate and level of channel coding for the uplink being relayed to the mobile station. Alternatively, the appropriate vocoder rate and level of channel coding for the downlink may be determined by the mobile station, and the appropriate vocoder rate and level of channel coding for the uplink may be determined by the base station.

BACKGROUND OF THE INVENTION

[0001] 1. Field of the Invention

[0002] The present invention relates generally to wireless communicationsystems. More particularly, the present invention relates to a wirelesscommunication system having an adaptive multi-rate (AMR) vocoder tomaximize the voice quality while minimizing the level of channel coding.

[0003] 2. Description of the Related Art

[0004] As the use of wireless communication systems become increasinglypopular, a variety of methods are being developed to increase the numberof mobile communication devices a system can simultaneously service. TheGlobal System for Mobile Communications (GSM), also referred to as theGroup Speciale Mobile, is one example of a wireless communication systemwhich is constantly being adapted to increase the number of simultaneoususers.

[0005] The GSM system is modeled after standards created by the EuropeanTelecommunications Standards Institute (ETSI) and operates between atelecommunication base station (BS) and a mobile station (MS) using apair of frequency bands in a frequency division duplex (FDD)configuration. The first frequency band occupies the frequency spectrumbetween 890 to 915 Megahertz (MHZ), and the second frequency bandoccupies the frequency spectrum between 935 to 960 MHZ. Typically, thefirst frequency range is used for the lower power transmissions from theMS to the BS, and the second frequency range is used for the higherpower transmission from the BS to the MS. Each frequency range isdivided into 125 channels with 200 Kilohertz (kHz) spaced carrierfrequencies.

[0006] The GSM communication system is a time-division-multiple-access(TDMA) system. In the GSM TDMA system, each carrier frequency is dividedinto eight (8) time slots. Because each MS is assigned a single timeslot on one channel in both the first frequency range and the secondfrequency range, a total of 992 MS may use the BS at the same time.

[0007] A typical speech channel for GSM communication is sampled at 8KHz and quantized to a resolution of 13 bits, providing for thedigitization of speech ranging from 0-4 KHz by a voice encoder, alsoreferred to as a vocoder. The 13 bits are then compressed by a factor ofeight (8) in a full-rate vocoder to a voice data digital bit stream of13 kilobits per second (Kbit/s). Because GSM uses a complex encryptiontechnique with interleaving and convolution coding, a high degree ofsystem integrity and bit error control are achieved. In fact, despitemulti-path and co-channel interference, the GSM system may continue tooperate despite a carrier-to-interference ratio (C/I) as low as 9 dB, incomparison to a typical advanced mobile phone service (AMPS) analogsystem requiring a maximum C/I of 17 dB.

[0008] Depending upon the topography of an area, a typical BS mayprovide communication services to any number of MSs within a radius upto 35 Kilometers. Consequently, with the rising popularity of mobilecommunication devices, it is often the case that during peak periods ofuse, such as rush-hour traffic, all channels are fully occupied and theBS is not able to provide communication services all of the MS in itsregion.

[0009] In order to avoid the inability to service all MS within aregion, the ETSI has contemplated a modification of the GSM standard toincrease the density of the 9 communication channels. However, becausethe allocated frequency spectrum of 25 MHZ with 125 separate 200 KHzcarrier channels is fixed, a current approach to increasing the densityof the communication system is to increase the number of users perchannel. In general, this density increase is achieved by decreasing theamount of digital information which is sent to and from each BS, therebyallowing each BS to support more users in a 200 kHz frequency band.

[0010] One approach to decreasing the amount of digital informationpassing between a BS and a MS is to decrease the vocoder rate of thedigital voice data from a full-rate vocoder rate of 13 kilobits persecond (Kbits/s) to a half-rate vocoder rate of 5.6 Kbits/s. Althoughthe ability currently exists to effectively double the number of userson any one communication channel from eight (8) to sixteen (16) by usingthe half-rate vocoder, it has been found that the 5.6 Kbits/s vocoderrate is barely acceptable as the speech quality is significantlydecreased.

[0011] In light of the above, it would be advantageous to provide acommunication system that provides for the user density of a half-ratevocoder system, while providing the voice quality approaching orexceeding that of a full-rate vocoder system. It would also beadvantageous to provide a communication system that provides for themodification of the communication channel to incorporate only the amountof channel coding necessary to achieve a reliable communication linkbetween the MS and the BS.

SUMMARY OF THE INVENTION

[0012] Broadly, the present invention provides for a wirelesscommunication system having the ability to increase or decrease thevocoder rate and channel coding in response to the level of interferencepresent on the wireless communication channel, resulting in acommunication channel having the best possible speech quality. This maybe accomplished in either a full-rate or half-rate GSM communicationsystem by decreasing the amount of channel coding during periods of lowchannel interference to allow transmission of more speech information,representing a higher vocoder rate and resulting in a higher speechquality. During periods of higher channel interference, the amount ofchannel coding may be increased to the maximum channel coding allowed ina GSM communication network. This increased channel coding provides forconsistent and reliable call handling, and results in a lower vocoderrate having a lower speech quality.

[0013] In an embodiment of the present invention, atime-division-multiple-access (TDMA) communication system includes abase station (BS) and at least one mobile station (MS), eachtransmitting and receiving an analog radio-frequency signal carryingdigitally coded speech. The speech is digitally encoded using a vocoderwhich samples a voice signal at different encoding rates. Alternatively,the speech may be encoded using a number of different vocoderssimultaneously, with each vocoder having a different encoding rate.During periods when the radio-frequency channel is experiencing highlevels of channel noise or interference, the encoded voice channelhaving a lower encoding rate is chosen. This lower-rate encoded voice iscombined with the high degree of channel coding necessary to ensurereliable transmission. When the radio-frequency channel is experiencinglow levels of channel interference, less channel coding is necessary andthe vocoder having a higher encoding rate is used. The high-rate encodedvoice is combined with the lower degree of channel coding necessary toensure reliable transmission. The appropriate level of channel codingnecessary for reliable transmission is determined by various channelmetrics, such as frame erase rate and bit error rate.

[0014] The determination of the appropriate vocoder rate and level ofchannel coding for both the uplink and downlink may be determinedcentrally at the base station, with the vocoder rate and level ofchannel coding for the uplink being relayed to the mobile station.Alternatively, the appropriate vocoder rate and level of channel codingfor the downlink may be determined by the mobile station, and theappropriate vocoder rate and level of channel coding for the uplink maybe determined by the base station.

BRIEF DESCRIPTION OF THE DRAWINGS

[0015] The nature, objects, and advantages of the invention will becomemore apparent to those skilled in the art after considering thefollowing detailed description in connection with the accompanyingdrawings, in which like reference numerals designate like partsthroughout, wherein:

[0016]FIG. 1 is a diagram of a typical wireless telecommunicationsystem, including a base station and a number of mobile stations;

[0017]FIG. 2 is a schematic diagram of the hardware of a typicalwireless transceiver of the present invention and includes threeseparate vocoders, each having a different vocoder rate;

[0018]FIG. 3 is a graph of the relative performance characteristics of awireless communication system implementing a variable vocoder rate;

[0019]FIG. 4 illustrates the coding, combination and interleaving ofspeech blocks into a frame, and the variation of the ratio of channelcoding to speech coding for various levels of radio-frequency channelnoise and interference;

[0020]FIG. 5 is a state diagram illustrating the change of vocoder ratebased upon the current status of the communication system, including theFE and BER metrics;

[0021]FIG. 6 depicts a sequence of steps which are performed in thecommunication system wherein the mobile station calculates the downlinkvocoder rate based on its calculations of a number of channel qualitymetrics;

[0022]FIG. 7 depicts a sequence of steps which are performed in thecommunication system wherein the mobile station forwards its channelquality metrics to the base station where the vocoder rate for thedownlink is determined and communicated to the mobile station;

[0023]FIG. 8 is a quantization table identifying the bits transmittedfrom the mobile station to the base station in order to provide the basestation with the necessary channel metric information to determine themobile stations vocoder rate;

[0024]FIG. 9 is a quantization table identifying the received bits whichcorrespond to the channel quality metrics made by the mobile station;and

[0025]FIG. 10 depicts a sequence of steps which are performed in thecommunication system wherein the base station calculates the uplinkvocoder rate based on its calculations of a number of channel qualitymetrics.

DETAILED DESCRIPTION

[0026] System Architecture of a Preferred Embodiment

[0027] Referring first to FIG. 1, an exemplary communication system ofthe present invention is shown and generally designated 100.Communication system 100 operates in compliance with the GSMcommunication standard which includes a time-division-multiple-access(TDMA) communication scheme. In general, a TDMA communication systemprovides for the transmission of two or more data channels over the sameradio-frequency channel by allocating separate time intervals for thetransmission of each data channel. In a GSM system, each 200 kilohertz(kHz) radio-frequency channel is divided into repeating time frames,each frame having a duration of 4.615 milliseconds. Each frame containseight (8) time intervals (also called “slots”) each having a duration of577 microseconds (4,615/8) and assigned to a different user.Communication system 100 includes a base station (BS) 102 which receivessignals from a mobile switching center (MSC) 106 via communicationchannel 108. This communication channel includes telephone and/ordigital information which may typically originate from land-basedtelephone systems. Base station 102 transmits information to, andreceives information from, mobile stations (MS) 110, 112, and 114 whichare within cell 120. Cell 120 is a geographical region within which allmobile stations communicate with the base station 102. Typically, thesecells range may have radii ranging from twenty-five (25) to thirty-five(35) kilometers, and may include such geographical disturbances such asbuildings 130 or mountains 132. As used herein, the term “information”shall be defined to include digital data, encrypted digital data,convolutionally coded, softcoded, and/or hard-coded data, digital bitsor a bit stream, or any other data type known in the art.

[0028] Because a GSM-based communication system operates with pairedfrequency bands in a frequency-division-duplex (FDD) mode, base station(BS) 102 sends information to the mobile station (MS) 110 over a firstradio-frequency channel 116, typically in the 890 to 915 MHZ range andreferred to as the “downlink,” and mobile station 110 sends informationto the base station 102 over a second radio-frequency channel 118,typically in the 935 to 960 MHZ range and referred to as the “uplink.”Although a GSM-based communication system operates using two frequencybands, it is nonetheless possible to implement the present invention ina system where both the BS and MS transmit and receive over the sameradio-frequency channel.

[0029] Communication system 100 may support a number of mobile stations(MSs) 110, 112, and 114. In fact, under the GSM standard, each 25 MHZfrequency band is divided into 125 channels with 200 Kilohertz (KHz)spaced carrier frequencies. With each carrier frequency supporting eight(8) separate users, a single GSM communication system may support nearlyone thousand (1,000) simultaneous users.

[0030] Given the high possible number of simultaneous users contributingto co-channel interference, and the presence of atmospheric andgeographical sources of interference, there are periods of time duringwhich a considerable amount of channel noise and interference is presenton the communication system 100. Moreover, the presence of buildings 130and mountains 132 result in multi-path distortion which further degradethe reliability of transmissions through the communication system 100.Additionally, because each MS may be moving in a different directionwith respect to the BS, either towards or away from the BS at speeds upto 250 kilometers-per-hour (156 miles-per-hour), the possibility that acommunication link will be temporarily or permanently disrupted is evenhigher.

[0031] In an attempt to minimize the deleterious effects of channelnoise and interference on both the uplink and downlink communicationchannels, a significant amount of channel coding is added to the digitalvoice data. Channel coding is generally defined to include the processof combining the encoded digital voice data from the vocoder, with anyredundant data, parity data, cyclic-redundant-checking (CRC) or othercheck data necessary to ensure the reliable transmission of the voicedata. The code rate is the ratio of data bits to total bits (k/n), andis typically just over one-half ({fraction (1/2)}) in an ordinaryGSM-based system with full-rate vocoders, and just below one-half({fraction (1/2)}) in a system with half-rate vocoders.

[0032] During the channel coding process, the introduction of thenecessary error correction, redundant data, parity data, CRC or checkdata is accomplished by convolutionally coding the digital voice datafrom the vocoder, with the necessary channel coding data. This resultsin a convolutionally encoded digital data stream which includes amixture of voice data and channel coding. As will be more thoroughlydiscussed in conjunction with FIG. 2, this digital data stream ismodulated and amplified for transmission over a radio-frequency channel.Upon reception of the modulated digital data stream, the data stream ischannel de-modulated and the voice data and channel coding isconvolutionally decoded and separated.

[0033] During periods with high levels of channel noise, theintroduction of significant channel coding provides for an increasedreliability of the communication channel. On the other hand, a datastream containing a significant amount of channel coding informationlimits the amount of voice data which can be transmitted, and duringperiods of low channel noise, results in an inefficient use of thecommunication channel. Consequently, the present invention monitors thecurrent level of channel noise, and either increases the amount ofchannel coding to improve channel reliability, or decreases the amountof channel coding to provide for the transmission of more voice data.

[0034] Although the current GSM-based communication systems dictate amaximum vocoder rate of 13 Kbits/s for a full-rate vocoder system, thepresent invention contemplates a much higher maximum vocoder rate up tothe bandwidth limitation of the wireless communication channel itself.For instance, if the current level of channel noise or interference isminimal, it is possible to provide a communication link having virtuallyno channel coding and thus providing for a voice data rate of 22Kbits/s. This would correspond to a voice bandwidth over 4 kHz,resulting in a voice channel having a frequency range and correspondingvoice quality well beyond that of a traditional 4 kHz voice bandwidth.

[0035] Transceiver Architecture

[0036] Referring now to FIG. 2, a circuit diagram of a transceiver ofone embodiment of the present invention is shown and generallydesignated 200. The transmitter portion of circuit 200 includes amicrophone element 202, such as an electret-type microphone, thatreceives an acoustic signal, such as a users voice, and converts theacoustic voice signal to an analog electrical signal. This analogelectrical signal passes through amplifier 204 for amplification andfiltering, and is supplied to the inputs of three (3) separate voiceencoders, or vocoders 206, 208, and 210.

[0037] A vocoder is an analog-to-digital converter (ADC) which iscreated especially for the digital encoding and compression of analogvoice data. Vocoders are designed around high speed digital signalprocessors (DSPs) and use a form of linear predictive coding which isintended to model the human vocal chords in order to produce realisticsynthetic speech with the minimum of memory. In a GSM communicationsystem with full-rate vocoders, voice data is sampled at the rate of 8kHz and quantized to a resolution of thirteen (13) bits and compressedto a bit rate of 13 Kbits/s. In a GSM communication system withhalf-rate vocoders, voice data is sampled at the rate of 8 kHz andquantized to a resolution of thirteen (13) bits and compressed to give abit rate of 5.6 Kbits/s.

[0038] In the present invention, vocoders 206, 208, and 210 eachreceives the amplified voice signal from amplifier 204 and each vocoderis continuously encoding the acoustic voice signals at different rates.For example, vocoder 206 may encode the voice signal at a vocoder rateof 8 Kbits/s, vocoder 208 may encode the voice signal at a lower vocoderrate of 6 Kbits/s, and vocoder 210 may encode the voice signal at aneven lower vocoder rate of 4 Kbits/s. The particular vocoder ratesdiscussed herein are merely exemplary, and it is to be appreciated thata vocoder of virtually any rate may be used, so long as therepresentative digital data rate is capable of being transmitted overthe radio-frequency communication channel.

[0039] The outputs from vocoders 206, 208 and 210 are fed into switch212 which is controlled by processor 214 having a memory storage 215.Processor 214 in the present embodiment is a microprocessor. However,processor 214 may instead be any conventional single or multi-chippedmicroprocessor, digital signal processor, microcontroller, or any othersuitable digital processing apparatus known in the art. Memory storage215 in the present embodiment may include an electrically erasableprogrammable read-only-memory (EEPROM), read-only-memory (ROM),random-access-memory (RAM), diskettes or other magnetic recording media,optical storage media, or any combination thereof. Electronicinstructions for controlling the operation of processor 214, in the formof program code, may be stored in memory storage 215.

[0040] Based upon a predefined selection process, described in greaterdetail below, processor 214 determines the proper vocoder rate andselects the output of the appropriate vocoder 206, 208 or 210 forpassage through switch 212 to encoder 216. For example, if the voicesignal is to be encoded at a full-rate of 8 Kbits/s, then the output ofvocoder 206 would be selected by processor 214 and passed through switch212. Alternatively, if the voice is to be encoded at the rate of 6Kbits/s, the output of vocoder 208 would be selected. Encoder 216receives the digital voice data from the vocoder and adds the level ofchannel coding corresponding to the vocoder rate selected.

[0041] Once passed through encoder 216, the now-encoded digital voicedata is mixed with the analog output of voltage-controlled-oscillator(VCO) 218 to modulate the digital voice data onto a carrier frequency inmodulator 220. Modulator 220 modulates a gaussian-minimum-shift-key(GMSK) signal on a radio-frequency carrier which is then passed throughvariable power amplifier 222 and through transmit/receive switch 224 toantenna 226 for transmission. A GMSK signal incorporates gaussian-shapedpulses and is intended improve the resilience of the communicationchannel to co-channel interference. As an alternative to GMSK, othermodulation methods known in the art may be used, such as BPSK, QPSK, orFSK.

[0042] Control of transmit/receive switch 224 is accomplished byprocessor 214 in a method well known in the art. In single antennatransceivers, it is often necessary to switch the antenna between thetransmitting and receiving portions of the circuitry in order to isolatethe sensitive receiver electronics from the higher power signalgenerated by the transmitter.

[0043] The receiver portion of circuit 200 begins with antenna 226 whichreceives an analog radio-frequency signal that is passed throughtransmit/receive switch 224 to intermediate-frequency (IF) amplifier andmixer 240. Mixer 240 removes the carrier frequency from theradio-frequency signal and passes the remaining analog signal to ananalog-to-digital converter (ADC) 242. ADC 242 converts the receivedanalog signal to a digital signal which is then passed throughequalization block 244 where the digital signal may be filtered and thedigital bit stream recovered, and to channel decoder 246.

[0044] As will be discussed in more detail below, processor 214 receivesa signal, in the form of rate bits, from channel decoder 246. These ratebits identify the appropriate vocoder rate needed to decode the currentvoice data encoded in the received signal. Based upon the rate bits,processor 214 selects vocoder 250, 252, or 254 using switch 248, and thedigital voice data from the decoded radio-frequency channel is passedfrom channel decoder 246, through switch 248, and to the appropriatevocoder 250, 252, or 254. For example, if the digital voice data wasencoded at a full-rate of 8 Kbits/s, then processor 214 would operateswitch 248 to send the digital voice data to vocoder 250 which, in thepresent embodiment, decodes at the full-rate of 8 Kbits/s. Vocoder 250decodes the digital information received from channel decoder 246, andre-creates the original analog voice signal which is then passed throughamplifier 256 and out speaker 258 to be heard by the user.

[0045] In an alternative embodiment, vocoders 206, 208 and 210 ofcircuit 200 may be replaced by a single vocoder (shown by dashed lines270) having multiple encoding rates, or a variable encoding rate. Forinstance, the single variable-rate vocoder 270 may be capable ofencoding acoustic signals from amplifier 204 at rates between 4 Kbits/s,and 8 Kbits/s as determined by processor 214. Similarly, vocoders 250,252, and 254 may be replaced by a single variable vocoder 272.

[0046] Although the present invention is discussed in conjunction with aTDMA communication system, it is to be appreciated that the use of aTDMA communication scheme is merely exemplary, and the present inventionmay be practiced on any number of alternative communications systems,such as code-division-multiple-access (CDMA) andfrequency-division-multiple-access (FDMA), for example.

[0047] Referring now to FIG. 3, a graphical representation of systemperformance is shown and generally designated 300. Graph 300 includes avertical axis 302 labeled “Voice Quality” and a horizontal axis 304labeled “Carrier-to-Noise Ratio (C/N).” As discussed herein, the termC/N is considered to include a carrier-to-interference (C/I) portion. Insummary, graph 300 represents the performance of communication systemsbased on the level of channel coding and corresponding vocoder rates.More specifically, three separate curves are shown and each representsthe performance of a particular system configuration. For example, graph306 represents the performance of a communication system using afull-rate vocoder, with a minimum level of channel coding. As can beenseen, curve 306 begins at a higher initial voice quality, but as the C/Ndecreases, the level of interference due to the lower level of channelcoding eventually causes a marked decrease in the voice quality.

[0048] Similarly, curve 312 represents the performance of acommunication system using a mid-rate vocoder with a correspondingmid-level of channel coding. Such a mid-rate vocoder rate could be 6Kbits/s. Although the initial voice quality shown by curve 312 ismaintained for a longer period, it too suffers from the interferencecaused by the lower level of channel coding.

[0049] Finally, curve 318 represents the system performance of acommunication system using a low-rate vocoder with a correspondinghigher level of channel coding. In this case, the high level of channelcoding provides for a continuous communication link despite asignificant decrease in the C/N, however, the voice quality is lowerthan either the system shown by curve 306 or 312.

[0050] In order to maintain the highest level of voice quality possible,despite the decreasing C/N, the present invention changes the voiceencoding rate and corresponding level of channel coding in order tomaximize the voice quality. For example, in environments where the C/Nratio is high, the system uses the highest possible vocoder rate andlowest possible amount of channel coding. In this situation, because ofthe low levels of noise and interference on the communication channel,there is little need for heavy channel coding to ensure thecommunication channel is sustained. However, as the C/N ratio begins todecrease, at the precise instant when curve 306 crosses curve 312, shownas intersection 310, the communication system of the present inventionchanges the vocoder rate and corresponding channel coding to the rateassociated with curve 312. In this manner, the highest possible level ofvoice quality is maintained, even though there is a higher level ofchannel coding present.

[0051] Similarly, as the voice quality of the system represented bycurve 312 drops to the level of the system represented by curve 318,shown at intersection 316, the communication system of the presentinvention again changes the vocoder rate and corresponding channelcoding to the rate associated with curve 318. In this manner, the voicequality for the communication channel is always maximized.

[0052] In the event the channel noise and interference exceeds themaximum allowable level and results in the voice quality beingsufficiently poor so as to pass below the threshold 324, shown atintersection 322, the communication channel is terminated. Thisterminated communication channel is perceived by the user as a “droppedcall.” Once terminated, system 100 must be re-initialized and acommunication channel must be re-established between the BS 102 and theMS 110.

[0053] Graph 300 has been divided into three (3) regions 308, 314, and320, representing the maximized voice quality. A communication channelusing the present invention will operate within each of these regions asneeded to maximize the voice quality. For example, for a communicationchannel which is initiated at a vocoder and channel coding rate inregion 314 corresponding to curve 312, a momentary decrease in the C/Nmay cause the system to switch to a vocoder and channel coding rate inregion 320 corresponding to curve 318. However, once the C/N returns toits original value, the system will shift back to the vocoder andchannel coding rate of region 314 corresponding to curve 312. In thismanner, system 100 may constantly move between regions 308, 314 and 320to maximize the voice quality of the communication channel.

[0054] Graph 300 has been shown to include three (3) separate curvesrepresenting three (3) different vocoder rates and corresponding levelsof channel coding. However, it should be appreciated that the selectionof three (3) vocoder rates is merely exemplary, and virtually any numberof vocoder rates may be used in the present invention. Moreover, in asystem of the present invention incorporating a vocoder having avariable vocoder rate, virtually any combination of vocoder rate andchannel coding may be accomplished within the system limits, rangingfrom a maximum vocoder rate with no channel coding, to minimum vocoderrate with maximum channel coding. Referring now to FIG. 4, adiagrammatic representation of the construction of a GSM communicationchannel is shown and generally designated 400. Representation 400includes a series of three (3) speech blocks 402, 404, and 406. Speechblock 402 includes a channel coding portion 408 and a voice data portion410. A speech block represents the digital information which has beengenerated by the vocoder 206, 208 or 210 and channel coder 216 ofcircuit 200. Accordingly, the digital information within a speech blockincludes both the voice data and channel coding which has beendetermined necessary for the reliable transmission of the information.While FIG. 4 identifies a channel coding portion 408 and a voice dataportion 410 as separate portions of speech block 402, it is to beappreciated that such identification is merely for discussion purposes,and the voice data is actually interleaved with the channel coding tocreate a data stream having 228 bits.

[0055] Speech blocks 402, 404 and 406 are each shown having differentratios of channel coding portions and voice data portions. Morespecifically, speech block 402 is shown having a larger proportion ofchannel coding 408 to a smaller proportion 412 of voice coding 410.Speech block 404, on the other hand, has approximately an evenproportion 418 of channel coding 414 to voice coding 416. Speech block406 has a larger proportion 424 of channel coding 420 to voice coding422. In any case, from comparing speech blocks 402, 404, and 406, it canbe seen that the ratios of channel coding to voice coding may change,and even though three separate ratios have been shown in FIG. 4,virtually any proportion 412, 418, and 424 may be implement with thepresent invention.

[0056] In addition to having a variable quantity of voice data andchannel coding, a speech block may also be encoded with a number of ratebits 413. These rate bits 413 represent the particular vocoder rate withwhich the voice data is encoded. For example, in a communication systemwhere vocoder rates may be varied, rate bits 413 provide the necessaryvocoder rate information to successfully decode the voice data. In apreferred embodiment, the rate bits are positioned within the speechblock 402, but are not convolutionally encoded with the voice data andchannel coding. Rather, the rate bits 413 are “soft-coded” into thespeech block 402 such that they can be extracted without the need forconvolutionally decoding the speech block. The term “extract” in thepresent context may include convolutionally decoding, soft-decoding,hard-decoding, or any other manner of retrieving the digital informationfrom the data stream known in the art.

[0057] The “soft-coding” of the rate bits may be accomplished by placinga series of bits within a particular location of the speech block. Forexample, rate bits 413 may be placed at bit positions 70, 71, and 72 ofspeech blocks 402, 404 and 406. By positioning the rate bits atconsistent locations within each of the speech blocks, it is notnecessary to decode the block to determine the value of the rate bits.Instead, the value of the bits in bit positions 70, 71 and 72 could bedetermined simply by scanning those bits in the serial bit stream.Additionally, it is possible to place the rate bits in more than onelocation within each speech-block, providing for a measure of errorcorrection. For example, rate bits 413 could occur in three separatelocations within speech block 402, allowing the averaging of the bitswithin the three separate locations in order to provide the bestapproximation of the rate bits despite any transmissions errors.

[0058] In a preferred embodiment, rate bits 413 may represent athree-bit binary value corresponding to eight distinct vocoder rates.Table 1 below identifies such a table of eight distinct vocoder ratesbased upon three rate bits. As can be seen from Table 1, the rate bits413 may be assigned any vocoder rate within the vocoder range of thecommunication system. TABLE 1 Rate Bits for Corresponding Vocoder RatesRate Bits Vocoder Rate (Level) Vocoder Rate (Kbits/s) 000 Level 1  3.0Kbits/s 001 Level 2  4.0 Kbits/s 010 Level 3  5.0 Kbits/s 011 Level 4 6.0 Kbits/s 100 Level 5  7.0 Kbits/s 101 Level 6  8.0 Kbits/s 110 Level7  9.0 Kbits/s 111 Level 8 10.0 Kbits/s

[0059] There are three (3) rate bits identified in Table 1, however, thenumber of rate bits may vary depending upon the total number of vocoderrates available. For instance, if only two rates are available, a singlebit would be needed, with a bit value of “0” indicating one rate, andthe bit value of “1” indicating the other rate. Similarly, if only fourrates were available, two rate bits would be needed, with the bit valuesof “00” indicating a first vocoder rate, bit values of “01” indicating asecond vocoder rate, bit values of “10” indicating a third vocoder rate,and bit values of “11” indicating a fourth vocoder rate.

[0060] Although Table 1 includes a series of eight (8) vocoder ratesspaced 1 Kbits/s apart, it is to be appreciated that it is not necessaryfor the vocoder rates to be evenly distributed. In fact, it would beadvantageous for the communication system of the present invention tohave a number of vocoder rates within the operating range mostfrequently experienced by the system. For example, if the communicationsystem noise and interference characteristics indicate that the vocoderrate would typically be 6 Kbits/s, then it might be advantageous toprovide several vocoder rates within the 5 to 7 Kbits/s region in orderto maximize the voice quality. In such an environment, a series of eight(8) vocoder rates might include the following vocoder rates: 4.0Kbits/s, 5.0 Kbits/s, 5.5 Kbits/s, 6.0 Kbits/s, 6.5 Kbits/s, 7.0Kbits/s, 8.0 Kbits/s, and 9.0 Kbits/s. Using these vocoder rates wouldallow the communication system to adjust the vocoder rate just slightlyin order to provide the finest possible voice quality during periods ofslight fluctuations in the channel noise and interference levels, whileretaining the ability to significantly change the vocoder rate forperiods of heavy channel noise and interference.

[0061] Once the voice data has been encoded with the necessary channelcoding, and any rate coding, to form speech blocks 402, 404, and 406,each speech block is divided into four (4) sub-blocks. For example, “A”speech block 402 is split into sub-blocks “A₁” 432, “A₂” 434, “A₃” 436,and “A₄” 438. Likewise, “B” speech block 404 is split into sub-blocks“B₁” 440, “B₂” 442, “B₃” 444, and “B₄” 446, and “C” speech block 406 issplit into sub-blocks “C₁” 448, “C₂” 450, “C₃” 452, and “C₄” 454. Inthis manner, the 228 bit data stream in the speech block is broken intofour (4) sub-blocks of 57 bits each.

[0062] Using a combination of sub-blocks, a multi-frame 476 isconstructed which includes a continual string of data frames, with eachdata frame having eight (8) time slots 478, 480 and 482. As shown bymapping lines 462 and 464 in FIG. 4, frame 478 is constructed from the“A₃” sub-block 436 and the “B₁” sub-block 440. This combination ofsub-blocks into frames 478, 480 and 482 is called “frame-interleaving”and is intended to create a more robust communication channel.

[0063] In addition to this frame-interleaving, the even bits withinframe 478 are comprised of the data bits of the “B₁” sub-block 440, andthe odd bits within frame 478 are comprised of data bits of the “A₃”sub-block 436. This even bit/odd bit combination is called“bit-interleaving” and results in the distribution of a single speechblock over four contiguous frames. This distribution provides for animproved fault tolerance for the communication system, and incircumstances where the noise level and interference level are high,results in a more resilient communication channel.

[0064] In addition to the combination of sub-blocks 438 and 442, frame480 is also encoded with communication system specific coding. Forexample, using the GSM-based communication system of FIG. 1, frame 480is encoded with three (3) leading “tail bits” 482, a first “encodedvoice” bit stream 484 of fifty-seven (57) bits, a single “flag” bit 486,a twenty-eight bit “training sequence” 488, a second “flag” bit 490, asecond “encoded voice” bit stream 492 of fifty-seven (57) bits, three(3) trailing “tail bits” 494, and an eight and one-quarter (8¼) bit“guard” period 496. The first and second “encoded voice” bit streams 484and 492 represent the encoded voice which was present in the “B₁”sub-block 440 and the “A₃” sub-block 436, which included both the voicedata, channel coding, and rate bits.

[0065] Because Doppler shift and multi-path echoes in system 100 canaffect the received signal quality, each TDMA frame must includetraining sequence 488, also called training bits. The receiver in system100 compares these training bits with a known training pattern, and fromthis deduces the transfer function of the propagation path. An adaptivefilter is then created within processor 214 to perform the inversetransfer function, thus canceling any unacceptable distortion. Thisadaptive filtering is well known in the art, and is thus not discussedin more detail here.

[0066] Because of the frame-interleaving and bit-interleaving employedin this GSM-based communication system 100, it is not possible to decodethe voice information without re-assembling the sub-blocks 432-454 fromsuccessive frames 478-482 in multi-frame 476. Consequently, it isnecessary for the digitally encoded voice information to be temporarilystored, such as by temporarily placing the encoded voice informationinto storage 215 of circuit 200. Once a sufficient number of frames hasbeen stored in memory storage 215, the sub-blocks are then re-assembledand the voice data is decoded from the re-constructed speech blocks,removing all channel coding, and sent through switch 248 to vocoders250, 252, and 254.

[0067] A full-rate GSM-based system would assign each time slot within aframe to a different user. For example, each of the eight (8) time slotswithin a frame would be assigned to eight (8) different users. In ahalf-rate GSM-based system, the frame and slot timing remains the same,but instead of a user being assigned a time slot in every frame, theuser is assigned a time slot in every other frame.

Operation

[0068] Communication Channel Metrics

[0069] The operation for the present invention includes the modificationof the vocoder rate and level of channel coding to provide the bestpossible voice quality, while ensuring a reliable communication channel.In order to determine the appropriate level of channel coding necessaryto provide reliable communication, a number of channel quality metricsare considered by the present invention. Defined generally, thesechannel quality metrics include characteristics of the communicationchannel which may be measured, and by continually measuring thesechannel quality metrics, an accurate evaluation of the channel qualitymay be made.

[0070] One channel metric used to evaluate the quality of thecommunication channel is the uncoded Bit Error Rate (BER). The uncodedBER of a communication channel is defined as the ratio of the number ofbits in a data stream which are improperly demodulated to the totalnumber of bits transmitted. In general, a bit error is caused when thenoise power level in a communication system becomes comparable to theenergy level in each bit transmitted. Consequently, in a system with asmall channel-to-noise ratio (C/N), bit errors are more likely.Conversely, in a system with a large channel-to-noise ratio, bit errorsare less likely. Thus, on a fundamental level, the rate of occurrence ofbit errors, or the BER, provides an overall system quality metric.

[0071] An additional metric which may be used to evaluate the quality ofa communication channel is the RX Quality (RXQ) indicator. The RXQindicator as generally known in the industry is assigned a value by thenetwork, indicating the quality of the received signal based upon thecurrent BER. Table 2 below includes values for a typicalnetwork-determined BER with corresponding RXQ values. This table,however, represents an average received quality, and not aninstantaneous RXQ value. TABLE 2 GSM Standards for RX Quality Metric RXQual Corresponding Bit Error Rate Range of Actual BER (%) 0 Below 0.2Below 0.1 1 0.2 to 0.4 0.26 to 0.30 2 0.4 to 0.8 0.51 to 0.64 3 0.8 to1.6 1.0 to 1.3 4 1.6 to 3.2 1.9 to 2.7 5 3.2 to 6.4 3.8 to 5.4 6 6.4 to12.8 7.6 to 11.0 7 above 12.8 above 15

[0072] The GSM standards for the RXQ of Table 1 is an average valuemeasured during a predefined period of time. However, because thepresent invention contemplates an immediate response to a decrease inthe RXQ value, it is necessary to determine the RXQ metric on ablock-by-block basis. This block-by-block calculation of RXQ′, forexample, would be made within the MS for the downlink, and within the BSfor the uplink.

[0073] In the present invention, an RXQ′ metric is defined and isdynamically measured by re-encoding the decoded voice data coming out ofthe convolutional decoder and comparing them against the received bits.The RXQ′ value represents the number of bits different between thereceived bits and the re-encoded bits per block. The RXQ′ consequentlyprovides a combined indication of bit error rate and receiver qualityfor each block.

[0074] Referring briefly to FIGS. 2 and 4, the determination of the RXQ′metric is accomplished by decoding the voice data from a speech block402 within a received frame 480, and re-coding the voice data forcomparison to the encoded received data. The determination of the RXQ′metric takes place within circuit 200 by receiving a transmitted frame480 and passing the frame through transmit/receive switch 224 tointermediate frequency (IF) amplifier and mixer 240, through ADC 242 andequalizer 244, to channel decoder 246. In channel decoder 246, the frame480 is decoded to the original speech block which is then passed tostorage 215 for later use. Following storage of the original speechblock, all channel coding is removed to recover the original voice datawhich may also be stored in storage 215, or passed on through switch 248to vocoders 250, 252 or 254 for conversion to audio.

[0075] Once the original voice data is recovered from channel decoder246, the now-decoded voice data is then re-encoded through aconvolutional coding process identical to that of the channel encoder216 to exactly re-create the original coded speech block. Thisre-encoding may be accomplished using channel decoder 246, or the voicedata may be passed through a separate channel coder 247. By comparingthe original speech block stored in storage 215 with the newly re-codedspeech block from channel coder 247, an estimated bit-error-rate may bedetermined. For example, by comparing the received speech block with there-coded speech block, the existence of any error-correction which hastaken place within channel decoder will become apparent. Consequently,this dynamic method of error detection is considerably more sensitivethan other estimates of the BER, and may be done on a block-by-blockbasis.

[0076] An additional metric, SRXQ, is defined as the weighed sum ofprior RXQ′ measurements. The SRXQ metric is intended to introduce somehistory into the vocoder rate decision making process based on thereceiver quality. In one embodiment, the RXQ′ measurements for the priorfive (5) blocks are considered in the SRXQ measurement. The prior RXQ′measurements are weighted in accordance with the following equation:

SRXQ=SUM(2^(K−1))(RXQ′(K+4));

[0077] where K=−4, −3, −2, −1, and 0, and where RXQ′(0) is the measuredvalue for the most recent block.

[0078] An alternative channel quality metric, Frame Erase (FE), may beused to determine the overall quality of the channel. The FE metricrepresents the number of frames which have been determined to becorrupted, and consequently not used in re-generating the original voicedata. In other words, the FE metric represents a count of the number offrames which have been erased per unit time. The decision to erase aframe may be made using a number of criterion. In a present embodiment,the determination to erase a frame is made based on thecyclic-redundancy-checking (CRC), also generally known as a “parity”check. Based on a CRC value which is decoded from the received frame, aframe is either used or discarded, avoiding the use of a frame which mayhave 0 been improperly decoded or otherwise corrupted.

[0079] System Operation

[0080] Referring now to FIG. 5, a state diagram is shown and generallydesignated 500. State diagram 500 represents the changes in vocoder andchannel coding rates in response to changes in the communication systemenvironment. For discussion purposes, it is assumed that thecommunication system is initially experiencing a high carrier-to-noiseratio (C/N), and thus the system is initially in state 502 having arelatively high vocoder rate of 8 Kbits/s, with a correspondingly lowlevel of channel coding. In other words, state 502 is used in low-noiseenvironments, such as where the carrier-to-interference ration (C/I)exceeds 19 dB, wherein the majority of digital information with a speechblock may be voice data. System 100 will remain in state 502 so long asthe FE metric remains at zero (0), as indicated by control path 508.This results in a communication system having a superior voice quality.

[0081] In the event that a frame is erased resulting in the FE metricbecoming non-zero, the BER is computed to determine whether it meets orexceeds a threshold value. In the present embodiment, this thresholdvalue is one percent (1%), meaning that if more than one bit out of atotal bit stream of one hundred (100) bits is erroneous, the thresholdis met or exceeded. Once the FE metric becomes non-zero and the BER isabove the one percent (1%) threshold, the system changes to state 504via control path 510.

[0082] State 504 is used in environments exhibiting moderate levels ofnoise and interference, and combines a mid-range vocoder rate of 6Kbits/s with a moderate level of channel coding. In the current example,the vocoder and channel coding rate will remain at the mid-range ofstate 504 so long as the BER is greater-than-or-equal-to one percent(1%), and less than five percent (5%) (1% £BER<5%). In this state,typically where the C/I is between 10 and 19 dB, the communicationchannel exhibits a reasonably good voice quality.

[0083] If after a period of time the communication environment improvesand the FE metric returns to zero (0) and the BER becomes less than onepercent (1%), the system returns to state 502 via control path 512. Onthe other hand, in the event the system environment becomes more noisyand the channel-to-noise ratio (C/N) becomes smaller, the FE metric willlikely increase. If the FE metric increases to equal or exceed 5, andthe BER metric is greater-than-or-equal-to 5 percent (5%), (FE>5 and 5%£BER) the system passes to state 506 via control path 516. In thisstate, a higher degree of channel coding is implemented resulting in acorresponding lower vocoder rate of 4 Kbits/s. According to control path520, the system will remain in state 506 so long as the BER isgreater-than-or-equal-to 5% (5% £BER), typically when the C/I is between4 to 10 dB.

[0084] When the system is in state 506, and the communicationenvironment improves causing the FE metric to decrease to zero (0) andthe BER metric to decrease to less than five percent (5%), than thecommunication system will change to state 504 according to control path518, thereby decreasing the level of channel coding and improving thevoice quality of the system.

[0085] In the event the communication system is in state 506 and the FEand BER metrics continue to increase, the communication system mayeventually discontinue the communication channel resulting in a “droppedcall.” In the present embodiment, the communication channel will bediscontinued when the FE and BER rates exceed 20 and ten percent (10%),respectively, for example.

[0086] In order to ensure the proper operation of the system of thepresent invention, it is necessary that the metrics evaluated fordetermination of the system control between various states 502, 504, and506 include a measure of hysteresis. For example, if no hysteresis wereto be included between states 502 and 504, it would be possible for thesystem to oscillate rapidly between the two states, resulting in avocoder rate and level of channel coding which varies from frame toframe. Although this continual vocoder change is possible with thesystem of the present invention, it is unnecessary and may result in aninefficient use of system resources.

[0087] The discussion of the various FE and BER values set forth aboveis intended as one example of a preferred embodiment having three (3)different vocoder and channel coding rates. The FE and BER values setforth are merely exemplary, and any number of alternative FE and BERvalues may be chosen and implemented. The threshold values for the FEand BER values may be treated as system parameters, and may change fordifferent vocoders. Also, FIG. 5 shows a state diagram with three (3)states, however, any number of states may be created within the presentinvention.

[0088] Mobile Station Control of Downlink Rate

[0089] Referring now to FIG. 6, a flow chart representing the operationof the communication system of the present invention is shown andgenerally designated 600. In general, this configuration includes the MSdetermining the proper downlink vocoder rate and level of channelcoding. Following this determination, the MS then transmits thenecessary rate information to the BS.

[0090] Flow chart 600 begins with first step 602 which includesreception of a radio-frequency frame at the MS. Following receipt of theframe at the MS, the soft-coded rate bits a re extracted from the framedata in step 604. In a preferred embodiment of the present invention andas discussed above in conjunction with FIG. 4, these soft-coded ratebits may include three (3) bits of rate information that can identify upto eight (8) different vocoder and channel coding rates. The frame datais then convolutionally decoded to yield the original speech block instep 605.

[0091] Using the appropriate vocoder and channel coding rate informationextracted in step 604, the speech block is then decoded to recreate theoriginal voice data in step 606. In this manner, the MS may receive aframe containing voice data encoded with virtually any vocoder rate, andthe frame may be successfully decoded to the original voice data becauseall relevant vocoder rate information is transmitted within the frame inthe form of soft-coded bits.

[0092] In order to provide the best possible voice communicationchannel, the MS determines the channel quality metrics discussed above,such as FE, BER and RXQ, in step 608. The MS also calculates the SRXQvalue in step 610 and, based upon the results of the measured andcalculated metrics, determines the vocoder rate for optimal voicequality in step 612. In a preferred embodiment of the present invention,the rate bits corresponding to the new vocoder and channel coding rateare determined from a look-up table. Once the vocoder and channel codingrate is determined, the MS transmits a frame with the new downlinkvocoder rate convolutionally coded into the frame in step 614. Uplink626 represents the transmission of a frame from the MS to the BS.

[0093] In step 616, the BS receives the frame containing theconvolutionally-coded downlink vocoder rate for the next downlinktransmission. Because it is not necessary to know the downlink vocoderrate in order to decode the uplink transmission, the downlink vocoderrate may be convolutionally encoded instead of soft-coded.

[0094] In step 618, the BS decodes the received frame from the MSyielding the new downlink vocoder rate bits. These vocoder rate bits areused to determine, using a look-up table or the like, the new downlinkvocoder rate. Using that newly determined vocoder rate, the BS encodesthe voice data in step 619 for transmission to the MS. In step 620, theBS transmits the frame containing the convolutionally encoded voice dataand coded downlink vocoder rate bits to the MS. Downlink 628 representsthe transmission of a frame from the BS to the MS.

[0095] Importantly, each downlink message includes as soft-coded bitsthe rate information related to the speech block. This is so becausethere exists a possibility that a frame may become corrupted and nolonger readable. This corruption may create a situation wherein the MSmay have transmitted a message frame in the uplink changing the downlinkvocoder rate, and that frame was not successfully received by the BS. Ifthis occurs, the MS would expect to receive a frame having a new vocoderrate, while the frame actually received would be encoded at the oldrate. Additionally, in circumstances involving discontinuoustransmissions (DTX), such as when the MS is not transmitting to savebattery power, the channel characteristics and corresponding vocoderrate information could change significantly between transmitted frames.Consequently, in order to avoid such miscommunication, each speech blockis soft-coded with the rate information necessary to decode the speechblock.

[0096] In a preferred embodiment of the present invention as shown inFIG. 6, steps within sequence 600 identified by bracket 622 areperformed within the MS, and steps within sequence 600 identified bybracket 624 are performed within the BS.

[0097] In any one cycle of uplink-downlink transmissions shown in FIG.6, both the BS and the MS will inform the other of the appropriatevocoder rates for the transmitted message. For example, in an uplinkframe containing convolutionally-coded rate bits for the next downlinkframe, soft-coded rate bits will be present which will tell the BS whatvocoder rate to use in decoding the uplink frame. Similarly, in adownlink frame containing convolutionally-coded rate bits for the nextuplink frame, soft-coded rate bits will be present which will tell theMS what vocoder rate to use in decoding that downlink frame.

[0098] In the communication system of the present invention, it has beentermed that vocoder rate bits which are not convolutionally encoded are“soft-coded” into the speech block, and the vocoder rate bits which areconvolutionally encoder are “hard-coded” into the speech block. As analternative terminology, the vocoder rate information which isconvolutionally encoded into a speech block could also be considered an“inside” rate, as the vocoder rate information is within theconvolutional coding. Vocoder rate information which is soft-coded intothe speech-block is considered an “outside” rate, as the vocoder rateinformation is outside the convolutional coding.

[0099] Base Station Control of Downlink

[0100] Referring now to FIG. 7, a flow chart representing the operationof an alternative embodiment of the communication system of the presentinvention is shown and generally designated 700. In general, thisconfiguration includes the MS monitoring a series of channel metrics andrelaying this metric information to the BS for determining the properdownlink vocoder rate and level of channel coding. Following thisdetermination, the BS then transmits the soft-coded rate bits to the MSwith the following frame.

[0101] In first step 702, the MS receives a frame with soft-coded ratebits. In step 703, the MS extracts the soft-coded rate bits from theframe, and using a look-up table or the like, determines the appropriatedownlink vocoder rate and level of channel coding. In step 704, usingthis rate information, the MS decodes the frame, yielding a speechblock. In step 706, the vocoders are set to the appropriate rate andthis speech block is decoded to re-create the original voice in thespeech block.

[0102] During the decoding process, the MS is determining the channelquality of the communication system. For example, quality metrics suchas FE and RXQ may be determined in step 708. Following the determinationof FE and RXQ, a quantized vocoder value is determined in step 710 whichreflects the current communication channel quality. Referring aheadbriefly to FIG. 8, a quantization table is shown and generallydesignated 800. Quantization table 800 includes both the FE metric 802and the RXQ metric 804 which are measured at the MS, and lists a numberof non-uniform quantization values for each. These RXQ′ values are themid-range of the transmitted quantization levels, and represent a rangeof RXQ′ metric values. Since the FE and RXQ′ are both associated withthe receiver performance, the quantization of RXQ′ is based on the valueof FE to effectively quantize RXQ′ into eight levels. By locating thecurrent measured values of both the FE and RXQ on the quantizationtable, a series of three (3) quantization bits are identified. Forinstance, for a FE value of 1 and a RXQ value of 25, quantization bits1-0-0 are selected. Once the quantization bits are selected, in step 712a frame is transmitted from the MS to the BS with the quantization bitsfully encoded in the speech block. Uplink 730 represents thetransmission of a frame from the MS to the BS.

[0103] In step 714, the frame is received at the BS with thequantization bits fully encoded. This frame is decoded in step 716 toyield the original quantization bits. Referring briefly to FIG. 9, aquantization table 900 is shown which provides a look-up table toreconstruct the FE and RXQ′ values from the received quantization bits.For example, for quantization bits 1-0-0, a FE metric value 904 of “1”and an RXQ metric value 906 of “22.” These metrics derived from thequantization bits are then used to calculate the SRXQ metric in step718. Based upon the quantization bits and the results of the SRXQcalculation, a new vocoder rate is determined in step 720 by the MS. Instep 722, voice data for the next speech block is encoded using the newvocoder rate, with the new vocoder rate bits being soft-coded into thespeech block resulting in a new frame. This new frame is thentransmitted from the BS to the MS in step 724. Downlink 732 representsthe transmission of a frame from the BS to the MS.

[0104] Base Station Control of Unlink

[0105] In addition to the rate bits which are exchanged between the MSand the BS to govern the downlink vocoder and channel coding rate, therate bits corresponding to the operation of the uplink are alsoexchanged. This is accomplished by the BS analyzing similar channelquality metrics which are used to determine the appropriate downlinkvocoder rate as discussed in conjunction with FIG. 6.

[0106] Referring now to FIG. 10, a flow chart representing the operationof an alternative embodiment of the communication system of the presentinvention is shown and generally designated 1000. In general, thisconfiguration includes the BS monitoring a series of channel metricsdetermines the proper uplink vocoder rate and level of channel coding.Following this determination, the MS then transmits the soft-coded ratebits to the BS with the following frame.

[0107] Flow chart 1000 begins with first step 1002 which includesreception of a radio-frequency frame at the BS. Following receipt of theframe at the BS, the soft-coded rate bits are extracted from the framedata in step 1004. In a preferred embodiment of the present inventionand as discussed above in conjunction with FIG. 4, these soft-coded ratebits may include three (3) bits of rate information that can identify upto eight (8) different vocoder and channel coding rates. The frame datais then convolutionally decoded to yield the original speech block instep 1006.

[0108] Using the appropriate vocoder and channel coding rate informationextracted in step 1004, the speech block is then decoded to recreate theoriginal voice data in step 1008. In this manner, the BS may receive aframe containing voice data encoded with virtually any vocoder rate, andthe frame may be successfully decoded to the original voice data becauseall relevant vocoder rate information is transmitted within the frame inthe form of soft-coded bits.

[0109] In order to provide the best possible voice communicationchannel, the BS determines the channel quality metrics discussed above,such as FE, BER and RXQ, in step 1010. The BS also calculates the SRXQvalue in step 1012 and, based upon the results of the measured andcalculated metrics, determines the vocoder rate for optimal voicequality in step 1014. In a preferred embodiment of the presentinvention, the rate bits corresponding to the new vocoder and channelcoding rate are determined from a look-up table. Once the vocoder andchannel coding rate is determined, the BS transmits a frame with the newuplink vocoder rate convolutionally coded into the frame in step 1016.Downlink 1018 represents the transmission of a frame from the BS to theMS.

[0110] In step 1020, the MS receives the frame containing theconvolutionally-coded uplink vocoder rate for the next downlinktransmission. Because it is not necessary to know the uplink vocoderrate in order to decode the uplink transmission, the uplink vocoder ratemay be convolutionally encoded instead of soft-coded.

[0111] In step 1022, the MS decodes the received frame from the BSyielding the new uplink vocoder rate bits. These vocoder rate bits areused to determine, using a look-up table or the like, the new uplinkvocoder rate. Using that newly determined vocoder rate, the MS encodesthe voice data in step 1024 for transmission to the BS. In step 1026,the MS transmits the frame containing the convolutionally encoded voicedata and soft-coded uplink vocoder rate bits to the BS. Uplink 1028represents the transmission of a frame from the MS to the BS.

[0112] Importantly, each uplink message includes as soft-coded bits therate information related to the speech block. This soft-coding enablesthe BS to properly decode the speech block without knowing in advancethe vocoder rate. This is particularly advantageous because there existsa possibility that a frame may become corrupted and no longer readable.This corruption may create a situation wherein the BS may havetransmitted a message frame in the downlink changing the uplink vocoderrate, and that frame was not successfully received by the MS. If thisoccurs, the BS would expect to receive a frame having a new vocoderrate, while the frame actually received would be encoded at the oldrate. Additionally, in circumstances involving discontinuoustransmissions (DTX), such as when the BS is not continuouslytransmitting, the channel characteristics and corresponding vocoder rateinformation could change significantly between transmitted frames.Consequently, in order to avoid such miscommunication, each speech blockis soft-coded with the rate information necessary to decode the speechblock.

[0113] In a preferred embodiment of the present invention as shown inFIG. 10, steps within sequence 1000 identified by bracket 1030 areperformed within the BS, and steps within sequence 1000 identified bybracket 1032 are performed within the MS.

[0114] System Performance

[0115] The communication system of the present invention provides forthe block and bit interleaving thereby minimizing the disruption to thecommunication link caused by channel noise, interference, and droppedframes. In addition to such redundancy, the vocoder rate informationwhich is either hard-coded within the frame or soft-coded outside theframe, may also be repetitive. Such repetition will further enhance theresilience of the communication system of the present invention.Redundancy of the vocoder rate information, or rate bits, may beaccomplished by repeating the bits in several locations within thespeech frame, as mentioned above in conjunction with FIG. 4.

[0116] Like traditional GSM-based communication systems, thecommunication system of the present invention provides for the transfer,or “hand-off,” of a MS from one BS to another BS in a different cell. Insuch a hand-off, it would not be necessary to provide the new BS withany special rate information via the communication link 108 as allnecessary vocoder rate information is presented in each frametransmitted from the MS.

[0117] The present invention may be implemented in either a full-rate orhalf-rate GSM-based communication system. The encoding and transmissionof the vocoder rate information between the BS and MS in both the fulland half-rate system would be identical.

[0118] In addition to the modification of the vocoder rate and channelcoding as discussed above, the power level of the transmissions may alsobe modified in order to provide the best possible voice quality. Forexample, in FIGS. 8 and 9, rate bits 806 and 902 may take intoconsideration, in addition to the FE and RXQ′ metrics, a metric relatedto the power level of the transmission. In such a situation, the BS mayadjust the vocoder rate′ and channel coding, while at the same timeadjusting the BS transmit power to minimize the BER or FE, resulting inbetter voice quality.

[0119] While the present invention has been discussed at length withrespect to the transmission of voice data between a BS and a MS, itshould be appreciated that any digital data may be communicated in asimilar manner. In fact, because other types of digital data may not bedependent upon the audio sampling rates, a much higher data rate may beachieved using the present invention, and is fully contemplated herein.

Other Embodiments

[0120] While there have been shown what are presently considered to bepreferred embodiments of the invention, it will be apparent to thoseskilled in the art that various changes and modifications can be madeherein without departing from the scope and spirit of the invention asdefined by the appended claims and their equivalents.

What is claimed is:
 1. A wireless communication system comprising: amobile station; and a base station which transmits a downlink signal tothe mobile station wherein the downlink signal includes a downlink rateand wherein the mobile station extracts the downlink rate from thedownlink signal and decodes the downlink signal using this extracteddownlink rate.
 2. The wireless communication system of claim 1, whereinthe downlink rate is determined by the mobile station using one or morechannel quality metrics.
 3. The wireless communication system of claim2, wherein the downlink rate is communicated to the base station by themobile station using one or more rate bits in a uplink signal. 4 Thewireless communication system of claim 3, wherein the one or more ratebits are soft-coded into the uplink signal.
 5. The wirelesscommunication system of claim 2, wherein determination of the downlinkrate comprises comparing one of the one or more channel quality metricsto a predefined value, and selecting a downlink rate corresponding tothe predefined value.
 6. The wireless communication system of claim 5,wherein the downlink rate is communicated from the base station to themobile station using one or more rate bits in the downlink signal. 7.The wireless communication system of claim 6, wherein the one or morerate bits are soft-coded into the downlink signal.
 8. The wirelesscommunication system of claim 2, wherein the one or more channel qualitymetrics includes a frame erase metric.
 9. The wireless communicationsystem of claim 2, wherein the one or more channel quality metricsincludes a receive quality metric.
 10. The wireless communication systemof claim 2, wherein the one or more channel quality metrics includes abit-err or-rate metric.
 11. The wireless communication system of claim2, wherein the one or more channel quality metrics includes an averagereceive quality metric.
 12. A wireless communication system comp rising:a mobile station which monitor s one or more downlink channel qualitymetrics, wherein the mobile station determines one or more quantizationbits corresponding to the one or more downlink channel quality metrics;and a base station which receives an uplink signal including the one ormore quantization bits from the mobile station, and wherein the basestation determines a downlink rate corresponding to the one or morequantization bits and transmits a downlink signal to the mobile stationusing the downlink rate.
 13. The wireless communication system of claim12, wherein the downlink signal includes the downlink rate.
 14. Thewireless communication system of claim 12, wherein determination of thedownlink rate comprises comparing one of the one or more channel qualitymetrics to a predefined value, and selecting a downlink ratecorresponding to the predefined value.
 15. The wireless communicationsystem of claim 12, wherein the downlink rate is communicated from thebase station to the mobile station using one or more rate bits in thedownlink signal.
 16. The wireless communication system of claim 15,wherein the one or more rate bits are soft-coded into the downlinksignal.
 17. The wireless communication system of claim 12, wherein theone or more channel quality metrics includes a frame erase metric. 18.The wireless communication system of claim 12, wherein the one or morechannel quality metrics includes a receive quality metric.
 19. Thewireless communication system of claim 12, wherein the one or morechannel quality metrics includes a bit-error-rate metric.
 20. Thewireless communication system of claim 12, wherein the one or morechannel quality metrics includes an average receive quality metric. 21.A wireless communication system comprising: a base station; and a mobilestation which transmits an uplink signal to the base station, whereinthe uplink signal includes an uplink rate and wherein the base stationextracts the uplink rate from the uplink signal and decodes the uplinksignal using this extracted uplink rate.
 22. The wireless communicationsystem of claim 21, wherein the uplink rate is determined in the basestation and communicated to the mobile station.
 23. The wirelesscommunication system of claim 21, wherein the uplink rate information isdetermined by the base station using one or more channel qualitymetrics.
 24. The wireless communication system of claim 23, wherein thechannel quality metrics includes a frame erase metric.
 25. The wirelesscommunication system of claim 23, wherein the channel quality metricsincludes a receive quality metric.
 26. The wireless communication systemof claim 23, wherein the channel quality metrics includes abit-error-rate metric.
 27. The wireless communication system of claim23, wherein the channel quality metrics includes an average receivequality metric.
 28. A method of maximizing voice quality in a wirelesscommunication system including a base station and a mobile station, themethod comprising the steps of: determining the quality of a downlinkcommunication channel in the mobile station using one or more channelquality metrics; determining a new downlink rate based on the channelquality metrics; transmitting the new downlink rate from the mobilestation to the base station in an uplink signal; extracting the newdownlink rate from the uplink signal; creating a downlink signal usingthe new downlink rate; encoding the new downlink rate into the downlinksignal; transmitting the downlink signal from the base station to themobile station; receiving the downlink signal at the mobile station;extracting the new downlink rate from the downlink signal; and decodingthe downlink signal using the extracted new downlink rate.
 29. A methodof maximizing voice quality in a wireless communication system includinga base station and a mobile station, the method comprising the steps of:determining the quality of a downlink communication channel in themobile station using one or more channel quality metrics; selecting oneor more quantization bits corresponding to the one or more channelquality metrics from a predetermined group of quantization bits;transmitting the quantization bits from the mobile station to the basestation in an uplink signal; extracting the quantization bits from theuplink signal in the base station; determining a downlink rate from theextracted quantization bits; creating a downlink signal using thedownlink rate; encoding the downlink rate into the downlink signal;transmitting the downlink signal from the base station to the mobilestation; receiving the downlink signal at the mobile station; extractingthe downlink rate from the downlink signal; and decoding the downlinksignal using the extracted downlink rate.
 30. A method of maximizingvoice quality in a wireless communication system including a basestation and a mobile station, the method comprising the steps of:determining the quality of an uplink communication channel in the basestation using one or more channel quality metrics; determining an uplinkrate based on the channel quality metrics; transmitting the uplink ratefrom the base station to the mobile station in a downlink signal;extracting the uplink rate from the downlink signal; creating an uplinksignal using the uplink rate; encoding the new uplink rate into theuplink signal; transmitting the uplink signal from the mobile station tothe base station; receiving the uplink signal at the base station;extracting the new uplink rate from the uplink signal; and decoding theuplink signal using the extracted uplink rate.
 31. A wirelesscommunication system comprising: means for determining the quality of adownlink communication channel in the mobile station using one or morechannel quality metrics; means for determining a new downlink rate basedon the channel quality metrics; means for transmitting the new downlinkrate from the mobile station to the base station in an uplink signal;means for extracting the new downlink rate from the uplink signal; meansfor creating a downlink signal using the new downlink rate; means forencoding the new downlink rate into the downlink signal; means fortransmitting the downlink signal from the base station to the mobilestation; means for receiving the downlink signal at the mobile station;means for extracting the new downlink rate from the downlink signal; andmeans for decoding the downlink signal using the extracted new downlinkrate.
 32. A wireless communication system comprising: means fordetermining the quality of a downlink communication channel in themobile station using one or more channel quality metrics; means forselecting one or more quantization bits corresponding to the one or morechannel quality metrics from a predetermined group of quantization bits;means for transmitting the quantization bits from the mobile station tothe base station in an uplink signal; means for extracting thequantization bits from the uplink signal in the base station; means fordetermining a downlink rate from the extracted quantization bits; meansfor creating a downlink signal using the downlink rate; means forencoding the downlink rate into the downlink signal; means fortransmitting the downlink signal from the base station to the mobilestation; means for receiving the downlink signal at the mobile station;means for extracting the downlink rate from the downlink signal; andmeans for decoding the downlink signal using the extracted downlinkrate.
 33. A wireless communication system comprising: means fordetermining the quality of an uplink communication channel in the basestation using one or more channel quality metrics; means for determiningan uplink rate based on the channel quality metrics; means fortransmitting the uplink rate from the base station to the mobile stationin a downlink signal; means for extracting the uplink rate from thedownlink signal; means for creating an uplink signal using the uplinkrate; means for encoding the new uplink rate into the uplink signal;means for transmitting the uplink signal from the mobile station to thebase station; means for receiving the uplink signal at the base station;means for extracting the new uplink rate from the uplink signal; andmeans for decoding the uplink signal using the extracted uplink rate.34. A wireless communication system, comprising: a base station; amobile station which receives a downlink signal from the base stationover a downlink, the downlink signal having a ratio of voice data tochannel coding; means for determining the quality of the downlink; andmeans for adjusting the ratio of the voice data to channel coding of thedownlink signal to improve the quality of the downlink.
 35. The wirelesscommunication system of claim 34, wherein the means for determining thequality of the downlink comprises one or more channel quality metrics.36. A wireless communication system, comprising: a mobile station; abase station which receives an uplink signal from the mobile stationover an uplink, the uplink signal having a ratio of voice data tochannel coding; means for determining the quality of the uplink; andmeans for adjusting the ratio of the voice data to channel coding of theuplink signal to improve the quality of the uplink.
 37. The wirelesscommunication system of claim 36, wherein the means for determining thequality of the uplink comprises one or more channel quality metrics.